Microphone apparatus and method with catch-up buffer

ABSTRACT

At a microphone, voice activity is detected in a data stream while simultaneously buffering audio data from the data stream to create buffered data. A signal is sent to a host indicating the positive detection of voice activity in the data stream. When an external clock signal is received from the host, the internal operation of the microphone is synchronized with the external clock signal. Buffered data stream is selectively sent through a first path, the first path including a buffer having a buffer delay time representing the time the first data stream takes to move through the buffer. The data stream is continuously sent through a second path as a real-time data stream, the second path not including the buffer, the real-time data stream beginning with the extended buffer data at a given instant in time. The buffered data stream and the real-time data stream are multiplexed onto a single data line and transmitting the multiplexed data stream to the host.

CROSS-REFERENCE TO RELATED APPLICATIONS

This application is a continuation of U.S. patent application Ser. No.14/797,310, filed Jul. 13, 2015, the entire contents of which areincorporated herein by reference.

TECHNICAL FIELD

This application relates to acoustic systems, and, more specifically toprocessing data in these audio systems.

BACKGROUND

Different types of acoustic devices have been used through the years.One type of device is a microphone and one type of microphone is amicroelectromechanical system (MEMS) microphone, including a MEMS diehaving a diaphragm and a back plate. The MEMS die is supported by asubstrate and enclosed by a housing (e.g., a cup or cover with walls). Aport may extend through the substrate (for a bottom port device) orthrough the top of the housing (for a top port device). In any case,sound energy traverses the port, moves the diaphragm and creates achanging potential of the back plate, which creates an electricalsignal. Microphones are deployed in various types of devices such aspersonal computers or cellular phones.

Digital microphones now exist that convert the analog data produced bythe sensor into digital data. The digital data is utilized by differentprocessing elements in the microphone to perform different sets offunctions such as acoustic activity detection. Acoustic activitydetection requires time to be performed in a reliable manner.Unfortunately, this time delay in detection incurs latency, which allowsreal-time data to pile or back-up thereby reducing the efficiency andperformance of the system. The latency further requires use of a bufferto store audio data, while the acoustic activity detection is made.

The problems of previous approaches have resulted in some userdissatisfaction with these previous approaches, specially the latencythat is incurred and that stays in the audio path impacting userexperience in voice recognition tasks.

BRIEF DESCRIPTION OF THE DRAWINGS

For a more complete understanding of the disclosure, reference should bemade to the following detailed description and accompanying drawingswherein:

FIG. 1 is a block diagram of a microphone;

FIG. 2 is a block diagram of a system of two microphones and a host;

FIG. 3 is a block diagram of a host;

FIGS. 4A and 4B illustrate a timing diagram of the operation of thesystems described herein according to various embodiments of the presentinvention;

FIG. 5 is a flow chart of the operation of the systems described herein;

FIG. 6 is a diagram showing one example of stitching;

FIG. 7 is a flow chart showing a stitching approach;

FIG. 8 is a time line and data diagram showing the stitching approach ofFIG. 7;

FIGS. 9A and 9 B are a flowchart of another stitching approach;

FIG. 10 is a time line and data diagram showing the stitching approachof FIGS. 9A and 9B.

Skilled artisans will appreciate that elements in the figures areillustrated for simplicity and clarity. It will be appreciated furtherthat certain actions and/or steps may be described or depicted in aparticular order of occurrence while those skilled in the art willunderstand that such specificity with respect to sequence is notactually required. It will also be understood that the terms andexpressions used herein have the ordinary meaning as is accorded to suchterms and expressions with respect to their corresponding respectiveareas of inquiry and study except where specific meanings have otherwisebeen set forth herein.

DETAILED DESCRIPTION

The present approaches allow a first microphone to be operated in a modehaving a real-time data path and a path that includes buffered data. Thepresent approaches utilize a host processing device that enables thebuffered audio data of the first microphone to catch up or recover thelatency as compared to the real-time or live audio data capture. Amongother things, this allows the use of a second microphone where thesecond microphone does not have a buffer. Consequently, any latencyissues associated with the first microphone are traversed.

In many of these embodiments and at a host processing device, bufferedpulse density modulation (PDM) data and real-time PDM data that has notbeen buffered is received from a first microphone. The buffered PDM dataand the real-time PDM data have the same data content but arediscontinuous with respect to the other when received at the hostprocessing device. The buffered PDM data is processed over a first timeinterval and the real-time PDM data is processed over a second timeinterval. The host processing device is operated so that the second timeinterval is less than the first time interval. The real-time PDM data isstitched to an end of the buffered PDM data. The stitching is effectiveto time align the buffered PDM data with respect to the real-time PDMdata to create an output data stream that is sequentially ordered intime. This allows the latency that is otherwise always present in thisclass of acoustic activity detection MEMS microphones to be transferredto the host device, where it can be easily recovered by faster thanreal-time processing.

In other aspects, second real-time data is received from a secondmicrophone, the second microphone not having a buffer. In some examples,the second real-time data is inserted into the output stream after theconclusion of the latency recovery mechanism described herein.

In other examples, the processing of the buffered PDM data comprisesdetermining an existence of a trigger word or phrase in the buffered PDMdata. In yet other examples, the buffered PDM data and the real-time PDMdata are decimated. In some examples, the buffered PDM data and thereal-time PDM data are received in a multiplexed format.

In others of these embodiments, a host processing device includes aninterface and a processor. The interface has an input and output, and isconfigured to receive buffered pulse density modulation (PDM) data andreal-time PDM data that has not been buffered from a first microphone atthe input. The buffered PDM data and the real-time PDM data have thesame data content but having a latency and being discontinuous withrespect to the other when received at the host processing device. Theprocessor is coupled to the interface, and the processor is configuredto process the buffered PDM data over a first time interval and processthe real-time PDM data over a second time interval. The processor isoperated so that the second time interval is less than the first timeinterval. The processor is configured to stitch the real-time PDM datato an end of the buffered PDM data. The stitching is effective tosynchronize the buffered PDM data with respect to the real-time PDM dataand to create an output data stream at the output.

Referring now to FIG. 1, a low power acoustic activity detection (AAD)microphone 100 is described. The microphone 100 includes a charge pump102, a transducer 104 (including a back plate and diaphragm), an inputbuffer 106 (with adjustable gain), a sigma delta modulator 108, adecimator 110, an Acoustic Activity Detection (AAD) module 112, acircular buffer 114, a first up converter 116, a second up converter118, a control block (processor) 120, an internal oscillator 122, and aclock detector 124.

The microphone 100 provides Voice Activity Detection (VAD) capabilitiesat ultra-low power. The AAD module 112 (including a (VAD) gain block)detects voice and voice-like activity. The circular buffer 114 receivesdata in real-time. In one aspect, the buffer may of sufficient size tohold 256 msec of audio. In another aspect, the buffer size may betrimable to sizes other than 256 msec. The charge pump 102 providescharge or energy to the transducer 104, and the transducer 104 convertsan acoustic signal into an analog signal, which is stored in the inputbuffer 106. The sigma delta modulator 108 converts the analog signalinto a pulse density modulation (PDM) signal, and the decimator 110converts the PDM signal into a pulse code modulation (PCM) signal. PCMdata has two paths: a first path through the circular buffer 114 toup-converter 118, and a second path for real-time data that flowsdirectly through up-converter 116.

The first up converter 116 and second up converter 118 convert PCM datainto PDM data. The control block (processor) 120 determines whentransmissions are made to a host. The internal oscillator 122 supplies aclock signal and the clock detector 124 determines whether an externalclock has been received from an external host via pin 134.

The AAD module 112 detects acoustic activity in a low power operatingmode of the microphone. The sensitivity of this block is partiallycontrolled through the input gain of this block. The VAD gain portion ofthe AAD module 112 in one aspect has a trimable gain. The AAD module 112monitors the incoming acoustic signals looking for voice-like signature,without the need for an external clock on clock pin 134 and thisoperation occurs in the aforementioned low power sensing mode. Upondetection of acoustic activity that meets the trigger requirements, themicrophone 100 asserts a SEL/STAT pin 130 to wake up the rest of thesystem in the signal chain. Further, the microphone 100 providesreal-time PDM data on DATA line 132 when a clock is made available onthe CLOCK line provided by the system after it wakes up. The buffer 114stores a previous amount of data (e.g., the previous 256 msec of data ora pre-set trimmed amount which may be different from 256 msec) generatedprior to the activity detection. Once a clock signal has been detectedon pin 134, the microphone 100 transmits the buffered data to a host viaDATA line 132. Data output may start at the same time as the SEL/STATline 130 indicates detection of voice. Alternatively, data output maystart after receiving an external clock via pin 134.

Referring now to FIG. 2, another example of a system with a catch-upbuffer is described. The system includes a first microphone 202, asecond microphone 204, and a host 206.

The first microphone 202 includes a transducer 222 (including, forexample, a diaphragm and back plate), a sigma delta converter 224, adecimator 226, a buffer 228, a first up-converter 230, a secondup-converter 231, a transmitter 232, a buffer control module 234, acontrol module 236, an Acoustic Activity Detection (AAD) module 238, andan internal clock 240.

The second microphone 204 includes a transducer, but does not include abuffer. In these regards, the second microphone 204 may be a microelectro mechanical system (MEMS) device that converts sound energy intoan electrical signal. The second microphone 204 may include a back plateand a diaphragm. Other examples of microphones are possible.

The host 206 is, in one example, a processing element such as a codec ordigital signal processor. The structure of the host 206 is describedwith respect to FIG. 3. The host 206 receives data streams (that may bemultiplexed over a PDM data line 280). The first data stream is from thebuffer 228 and the second data stream is un-buffered data. The buffer228 introduces latency (delay), but is needed because the firstmicrophone 202 needs time for the AAD module 238 to determine whetherthere is voice (or other acoustic) activity. Additionally, the hostprocessor requires time to wake up from a low power mode and be ready toreceive data. The buffer also provides important contextual informationto a speech trigger recognition engine to allow it to perform better innoisy conditions. Because of the delay and latency, the two data streams(of the same data content) will be discontinuous and time delayed withrespect to each other. The host 206 operates to synchronize the two datastreams at its output, and eliminates any discontinuous aspects withrespect to each other. In other words, the host guarantees that at somepoint in time, input data that it is receiving (from one or both of thefirst or second microphones) is the same data that it is outputting.

The transducer 222 (which may be a micro electro mechanical system(MEMS) device) converts sound energy into an analog electrical signal.The sigma delta converter 224 converts the analog electrical signal intoa pulse density modulation (PDM signal). The decimator 226 converts thePDM signal into a pulse code modulation (PCM) signal. The buffer 228stores the PCM signals. The up-converter 230 converts PCM signals intoPDM signals. The transmitter 232 transmits a multiplexed signal (of thefirst and second data streams) over the data line 280. The transmissionis initiated with the receipt of the external clock on line 284. Thebuffer contents are monitored by the buffer control module 234. When thebuffer has transmitted the pre-determined amount of data, for example256 msec and some additional extension data (by “extension data” it ismeant as data beyond the buffer length), the buffer control module 234sends a buffer empty (bempty) signal 285 to the control module 236,which causes the transmitter 232 to stop multiplexing the contents ofthe buffer 228. The AAD module 238 detects whether there is voice orother acoustic signals and sends a SEL/STAT signal 282 when acousticactivity is detected. The host 206 responds with a clock signal 284,which is sent to the first and second microphones 202 and 204. Thesecond microphone 204 is also controlled via the GPIO 286 which keepsmicrophone 204 disabled. The effect of the clock signal 284 is to causemicrophone 202 to transmit data. A GPIO 286 is used to control power tothe second microphone 204 and to select the second microphone 204. TheGPIO 286 is asserted only after stitching is completed at the host. Theterm “stitching,” means combining the real-time data stream at the endof the buffered data stream in the host, such that a continuous datastream is presented to the application.

In one example of the operation of the system of FIG. 2, the firstmicrophone 202 stores or buffers data in the buffer 228 in order foracoustic activity detection to be performed by AAD module 238 on thebuffered data. The host 206 is awaken by the SEL/STAT signal 282 andresponsively sends the clock signal 284 to the first microphone 202.Receipt of the clock signal allows the first microphone 202 to clockdata out over data line 280.

The first microphone 202 sends multiplexed data (of the first and secondstreams) to the host 206. This multiplexed data will include real-timeand buffered data of length X time units (e.g., 256 ms).

The host 206 processes the X units of buffer data until the processingis complete. X units of real-time data is also waiting for processing bythe host 206. The host 206 processes the real-time data over a secondtime period that is much, much less than the first time period. The host206 may be operated faster to accomplish this function. The host 206stitches the real-time data to the end of the buffered data. The goal isthat the data being input into the host 206 is being output from thehost 206 in real-time.

In order to support low power applications that require or prefer toreduce the signal latency due to the buffer 228, a burst mode isprovided in the system of FIG. 2. Burst mode provides the capability forfaster than real-time data transfer. Burst mode implements two datapaths, one for the real-time data and the other for the buffered data,both of which go through the decimation and interpolation functionsneeded to run the AAD module 238, for example, at 16 kHz/16 bits PCM. Inone aspect, the burst mode utilizes two interpolators to ensure that thesignal paths for both signals have the same phase response, excludingany coding and decoding associated with the buffering operation.

The burst mode operates as follows. The SEL/STAT line 282 is used forsignaling the state of the microphone 202 to the host 206. Themicrophone 202 is normally in sense mode with no activity on the dataline 280 and SEL/STAT line 282, when there is no voice and themicrophone AAD module 238 has converged to the ambient noise.

When the AAD module 238 detects acoustic activity and asserts theSEL/STAT line 282, the host 206 enters the wake-up mode. This actionwakes up the host 206 with some latency. The host 206 in one aspectprovides a 768 kHz signal to the clock line 284.

The reception of the clock signal 284 by the first microphone 202 alongwith acoustic detection puts the first microphone 202 into burst mode.In one example, the first microphone 202 enters burst mode within 10clock cycles of receiving the external clock at 768 kHz. The burst modeuses a first PDM channel to send the buffer data and a second PDMchannel to send real-time data to the host.

In some aspects, the real-time PDM channel may be the default channel,so that the real-time data is valid and may be latched during the risingedge of the clock. Buffered data is valid and may be latched during thefalling edge of the clock. The data transfer rate in burst mode is inone example double the normal data rate at 768 kHz. When in the burstmode and in one example, the first microphone 202 will toggle theSEL/STAT pin 282 at 8 kHz, synchronous to the 768 kHz CLOCK edges. Whenthe buffer 228 is emptied via the burst mode, the SEL/STAT pin 282 isheld high so the host 206 is signaled that the first microphone 202 isnow caught up with real-time data. The host 206 may also use a count ofthe toggle to verify the amount of data collected to aid in “stitching”the buffered and real-time data. Slower toggle rates will cause loweroverhead on host systems. In one aspect, the use of an 8 kHz toggle ratewill allow the time between each transition to be the duration of 1 PCMsample.

The signal processing algorithms for decimation may cause pops or clicksat the stitch point of the real-time and buffered audio. By a “pop” or“click,” it is meant that unnatural discontinuities in the audio sampleswill cause distortions in the output audio signal that resemble a “pop”or “click” sound. Some overlap is expected to be required between thebuffered and real-time data to eliminate these pops or clicks. Thebuffered data will be extended beyond the 256 msec or the specifictrimmed size to provide this overlap. During the extended buffer state,the SEL/STAT line 282 is held high. The end of the extended bufferperiod is signaled by toggling SEL/STAT pin 282 at 16 kHz to allowdistinction from the burst mode state.

At the end of the extended buffer period or state, the first microphone202 enters the Real-Time low power mode. When in Real-Time low powermode, the first microphone 202 only uses one of the PDM channels. Datais valid during the rising edge. This permits the use of the second PDMmicrophone 204 on the PDM port. The second PDM microphone 204 has to beoff during the combined time for burst mode output and extended bufferoutput durations. The SEL/STAT toggle on line 282 may be used as asignal to determine when the second microphone 204 can be powered on.The SEL/STAT pin 282 will keep toggling until the end of detected voiceactivity. Thus, the activity of the SEL/STAT pin 282, either high ortoggling is an indicator of voice activity. If the host 206 usesinternal timers available to it, exact grabbing of the extension buffermay not be necessary, but may be self-regulated by the host 206.

Only after the cessation of voice activity and the external clock 284from the host 206 will the first microphone 202 re-enter sense mode.

Referring now to FIG. 3, one example of a host 300 (e.g., host 206 fromFIG. 2) is described. The host 300 includes a stereo decimator 302(acting as an interface) and a processor 304. The decimator 302 convertsPDM data into PCM data. The processor 304 implements or executesstitching approaches (any combination of hardware and software) thatappend real-time data to the buffered data. The processor 304 includes abuffer for real-time data.

Data discontinuity exists at the start of a burst when the microphone(e.g., microphone 202) is operated in burst mode. Discontinuity can berepresented as x(m)-x(n) and is approximately equal to 256 ms where 256ms is the buffer length of the first microphone (e.g., microphone 202).A voice trigger algorithm starts recognition on the buffered data, x(m)over a first processing interval, while the real-time data x(n) is savedin a buffer on the host 300 and will be processed by voice triggeralgorithm over a second processing interval. Data is stitched by thehost 300 (e.g., host 206) after the entire buffer (256 ms) is drainedand latency is consequently recovered. Buffer data of the buffer in thefirst microphone (e.g., buffer 228 in first microphone 202) is extended(e.g., by a length less than 256 ms) to allow the stitch algorithmoperated by the processor 304 to synchronize x(m) and x(n) and eliminatesignal discontinuity.

After data discontinuity is resolved and synchronization is achieved,real-time data from the first and second microphones (e.g., microphones202 and 204) can be multiplexed on the incoming data line and output inreal-time. This may correspond to a low-power real-time mode.

Referring now to FIGS. 4A and 4B, a timeline showing the operation ofthe approaches described herein is described. The time line shows theoccurrence of voice activity 402. It will be appreciated that thistiming diagram illustrates the operation of the systems described withrespect to FIGS. 1-3.

Voice is detected causing the SEL/STAT line 404 to go high. SEL/STATstays high until the clock (e.g., 768 kHz clock) is received from thehost. The host sends clock signal 406 back to the first microphone. Thefirst microphone detects the clock signal and sends data out on dataline 408 at time 410. SEL/STAT then toggles at a suitably chosensignaling frequency. An example frequency that may be used is 8 kHz. Onthe rising edge of the clock, real-time PDM data 440 is received overthe data line. On the falling edge, buffer PDM data is received over thedata line from the first microphone. This is the burst mode.

Then at time 412, extension mode is entered. On the rising edge of theclock real-time PDM data is received over the data line and on thefalling edge of the clock extension buffer data is received over thedata line. This allows the host to stitch the real-time data to thebuffer data. The extension period may last a pre-determined time. In oneexample, this extension period is less than 128 ms and in otherexamples, this extension period could be 32 msec, 16 msec or 8 msec oranother suitable time interval. SEL/STAT toggles at a suitably chosensignaling frequency different from the burst mode signaling frequencyuntil AAD goes inactive. An example frequency could be 16 kHz. At thispoint, real-time PDM data alone is being received over the data line.Optionally, at time 414, a second microphone (without a buffer) may bepowered on the falling edge of the clock after the buffer extensionperiod. On the rising edge of the clock real-time PDM data from firstmicrophone is received over the data line and on the falling edge of theclock real-time PDM data from second microphone is received over thedata line.

Referring now to FIG. 5, one example of the state transitions isdescribed. It will be appreciated that this flow chart illustrates theoperation of the systems described with respect to FIGS. 1-4.

At step 502, the system is powered ON. At step 504, determine if theSEL/STAT line is VDD or floated.

If at step 504 VDD and acoustic activity detection (AAD) is off, then atstep 506 the external clock rate is determined. In one aspect of theinvention, if the clock rate is 0-315 kHz, at step 508, the microphonegoes to sleep mode. If the clock rate is between 315 and 1400 kHz, atstep 510, the microphone is operated in low power mode. If the clockrate is between 1.4 to 4.8 MHz, the microphone goes to normal operatingmode at step 512.

If at step 504 the SEL/STAT is floated, then at step 514 it isdetermined if there is an external clock being received at themicrophone. If the external clock is detected to be 1 to 4.8 MHz,execution continues with step 526 where the microphone is operated innormal operating mode. If the external clock is at 768 kHz, executioncontinues with step 524 at a low power real-time mode. If the answer atstep 514 is negative, at step 516 the microphone enters PDM sensingmode. At step 518, wake up is performed. If no external clock is beingreceived at the microphone, execution continues with step 516. Ifexternal clock is being received at the microphone, burst mode isentered at step 520. At step 520, burst mode is executed as has beendescribed herein. If at step 524 or step 526, the external clock isstopped, then the execution reverts to block 516 and the microphoneenters the PDM sensing mode.

Referring now to FIG. 6, one example of stitching data from the bufferand real-time data is described. It will be appreciated that thisexample shows how data may be stitched together in the system of FIGS.1-5.

A buffer (e.g., the buffer 228 in the first microphone of FIG. 2)includes buffered data 602. An audio phrase is received. “He” which (inthis example) is the first part of the phrase “Hello VT.” Real-time data604 also is received and this may be “llo VT” from the last part of thephrase “Hello VT.” The stitching algorithm in the host (e.g., host 206)receives these two data streams and stitches “llo VT” to the end of thebuffered data to make stitched data 606 “Hello VT.” The processing ofthe buffer data must proceed at a real-time rate as it is received at areal-time rate with the latency determined by the buffer size in themicrophone. The processing of the real-time data may be made much fasterthan real-time, because of the accumulated data in the host after thestitching process is completed. Thus, the stitched continuous datastream present at the output of the host recovers the latency andcatches up to the live signal with significantly reduced latency. Thebuffered data 602 and the real-time data 604 are now orderedsequentially with each other and the host can process the data receivedfrom one or more microphones in real-time without needing to considersynchronization issues between the first and the second microphone.

Referring now to FIG. 7 and FIG. 8, one example of a stitching approachis described. The discussion with respect to FIG. 7 and FIG. 8 assumes amicrophone and host device, for example, as described previously above.

Transients occur whenever PDM data is fed into a decimation filter orwhen it is stopped. In some aspects, when buffered data is followed bythe real-time data, the transients will occur in the middle of thecombined audio streams. Using an extended buffer of length greater thanthe end transient of the buffered audio and the start transient of thereal-time audio allows the skipping of these time intervals bycalculation of the decimation filter characteristics. One stitchingapproach provides an extended buffer and skips these transients. Thus,first the buffered and real-time signal must be time aligned at thehost. This is possible because both streams start simultaneously onlyafter the host clock is received.

The lengths of the buffer and the extended buffer are pre-determined andmay be based upon various factors. 256 ms and 16 ms are examples oflengths for the buffer and extended buffer, respectively.

The output data is taken from the buffered audio until it is past thepoint where the start transient of the real-time audio has damped out.The output data is then switched to the corresponding real-time stream,so that the transient at the end of the extended buffer data may beskipped. This audio stream does not have any transient in the middle ofthe stream with this stitching strategy.

At step 702, the host is asleep. At step 704, the microphone wakes upthe host, for instance, as has been described above.

At step 706, various parameters or variables are initialized. Morespecifically, Bufl is the length of the buffer and this is initializedto a value, for example, in milliseconds (e.g., 256 ms). Bufl is shownas element 802 in FIG. 8.

Stpt is the stitch point and is a time value as measured from the end ofBUFFERPCM. It is also the same time value when measured from thebeginning of the RT_BUF, the real-time buffer on the host. Stpt isrepresented as element 804 in FIG. 8. Extl is the length of theextension buffer in the microphone and is represented by element 803.

Rt_Buf[BufL+StPt] is an amount of allocation of memory space forreal-time data in the host. Real-time data will be stored in a real-timebuffer in the host. In one example, the real-time buffer Rt_buf could beset to 256 ms+8 ms if 8 ms is the stitch point. pWR and pRD are writeand read pointers and these are initialized to zero.

At step 708, a check is made to determine if line 130 (of FIG. 1) isactive. If it is not, return to step 702.

If the line is active at step 710, the host inputs the 2 channels(stereo) of data. The host decimates the data from PDM format to PCMformat.

At step 712, store the real-time PCM data in a real-time buffer usingthe pWR pointer to point to the correct place in the buffer to write thedata.

At step 714, a check is made to determine if the pWR pointer has gonepast the stitch point. If it has not, at step 716 output the buffereddata stream (buffered PCM data) so that it can be further processed. Atstep 718, the pWR pointer is incremented.

If at step 714, the pWR pointer has gone beyond the stitch point,control continues to step 720. A check is made to see if the pRD flag(used as a position pointer in the real-time data buffer in the host)has reached the stitch point. If it has, output real-time data at step726. If it has not reached the stitch point, real-time buffer data[pRD+StPt] is output. Then, the pRD pointer is incremented at step 724.

It can be seen in FIG. 8 that the output of this approach will haveregion 830 (from buffered PCM), region 832 (from extended buffer),region 834 (from extended buffer from RT buffer), region 836 (from RTbuffer), and region 838 (not from RT buffer), as the data comes in tothe host. It is apparent that the transient regions are avoided becausethe regions that include the transients are not used (as from aparticular buffer).

Referring now to FIGS. 9A and 9B and FIG. 10, another example of astitching approach is described. The discussion with respect to FIGS.9A, 9B and FIG. 10 assumes a microphone and host device, for example, asdescribed previously above. At step 902 in FIG. 9A, the host is asleep.At step 904, the microphone wakes up the host.

At step 906, various parameters or variables are initialized. Bufl isthe length of the buffer and this is initialized to a value, forexample, in milliseconds. Bufl is shown as element 1001 in FIG. 10. Trptis the length of decimator transients and these are represented byelements 1002 and 1004 in FIG. 10. Rt_Buf[BufL+TrPt] is a memoryallocation for real-time data. Real-time data will be stored in areal-time buffer in the host. This could be 256 ms+8 ms if 8 ms is thestitch point. pWR and pRD are write and read pointers and these areinitialized to zero. Extl is the length of the extension buffer in themicrophone and is represented by element 1003. IS is the interpolatedsample.

At step 908, a check is made by the host to see if line 130 of FIG. 1 isactive. If it is not, a return to step 902. At step 910, if it isactive, then input the 2 channels (stereo) of data is made. The data isdecimated from PDM format to PCM format.

At step 914, the approach is dealing with transient period lengths TrPt1002 and 1004 lengths, which are assumed to be equal. A check is made tosee if pWR is in that area of data.

If the pWR pointer is not in the transient area, at step 916 bufferedPCM data is output from the host and at step 918 pWR (which is thepointer used in the buffer to store real-time data in the host) isincremented.

If the approach has reached the transient portion, pWR is somewhere inthe middle of zone 1006. At step 920, a check is made to see if pWR isout of that zone 1006. If the answer is negative, then at step 922interpolate the output data based on weighting. PCM data that isinterpolated is output from the host at step 924. pWR and pRD areincremented at step 926.

If the determination made is that the pointers are out of the 1006 zone,then control continues with step 928 where a determination is made as towhether pRD is out of zone 1004. If not out of zone 1004, at step 930output real-time buffer data RT_BUF[pRD+TrPt]. At step 932, the pointerpRD is incremented.

If the process moves out of zone 1004 (by the determination at step928), real-time (unbuffered) data is output from the host at step 934.

It can be seen that an interpolated region in the output steam avoidsthe transients. The output is a buffered PCM data region 1030;interpolated region 1032 (that avoids the transients of regions 1002 and1004); and real-time buffer region 1034 (from the real-time buffer); andregion 1036, which is real-time data that is unbuffered.

It will be understood that different interpolation approaches may beused. If infinite input response (IIR) filters are used in decimation,then the transient persists in perpetuity though with decreasing energyto meet design goals. In some situations, the stitch point still showssome broadband noise at the stitch point when basic stitching is used.In interpolated stitching, an allowance is made for the most significantenergy of the transients to die down. Then, the intermediate timeinterval is used to linearly interpolate between the buffered andreal-time data. The interpolation may be performed in one example asfollows.

Let the time interval be given by discrete variable n. The start of thebuffered audio may be considered n=0. An assumption may be made that thetime for the most significant energy of the transients to die down isTrPt. The output for each section is given by the following equationsrespectively.

For the first segment 1030:op(n)=BUFPCM(n) for 0<n≦(BufL+TrPt)

This equation describes that the output of the host is determined solelybased on buffered data.

For the intermediate segment 1032:

op(n) = α(n) × ExtL(n) + [1 − α(n)] × RTBUF(n)for  (BufL + TrPt) < n ≤ (BufL + ExtL − TrPt)where  α(n) = n/(ExtL − 2 × TrPt)

This equation describes that data in the intermediate segment islinearly interpolated in both data streams.

For the segment 1034:op(n)=RTBUF(n) for(BufL+ExtL−TrPt)<n

This equation describes that the output of the host is determined solelybased on real-time buffered data. The above approach results insignificantly lower transient broadband energy in the segment where theoutput is in transition from the buffered data stream to the real-timedata stream.

In the equation above, op(n) is output at processing cycle n, n iscounter of processing cycles, BUFPCM(n) is buffer PCM sample ofprocessing cycle n, RTBUF(n) is real-time PCM sample of processing cyclen, ExtL(n) is extension buffer PCM sample of processing cycle n, andα(n) is time varying weight factor of processing cycle n. In one aspect,α(n) is defined to increase linearly from 0 to 1 with increasing n.

The first and last equations determine when the output is determinedsolely by the Buffered data and the Real-Time data and the intermediateequation determine how the data in the intermediate segment is linearlyinterpolated from both data streams.

This results in significantly lower transient broadband energy in thesegment where the output transitions from the buffered data stream tothe real-time data stream. In other words, buffered data is used more atthe beginning of the interpolation, while real-time data is used less.Real-time data is used less at the beginning and more at the end. Thedegree of use for each may be described as a linear function.

Preferred embodiments are described herein, including the best modeknown to the inventors. It should be understood that the illustratedembodiments are exemplary only, and should not be taken as limiting thescope of the appended claims.

What is claimed is:
 1. A method in a microphone apparatus, the methodcomprising: clocking the microphone apparatus using a local oscillatorof the microphone; generating a digital data stream from an electricalsignal representative of acoustic energy; determining whether voiceactivity is present in the electrical signal while buffering datarepresentative of the electrical signal in a buffer of the microphoneapparatus, the buffered data having a delay time with respect toreal-time data representative of the electrical signal; providing amultiplexed data stream at an interface of the microphone apparatusafter determining the presence of voice activity, the multiplexed datastream including a buffered data stream and a real-time data stream, thebuffered data stream and the real-time data stream based on theelectrical signal, the buffered data stream including non-real-timebuffered data and real-time buffered data, an end portion of thebuffered data stream temporally overlapping a beginning portion of thereal-time data stream; and providing the real-time data stream at theinterface after discontinuing providing the buffered data stream at theinterface.
 2. The method of claim 1, further comprising discontinuingproviding the multiplexed data stream at the interface of the microphoneapparatus after a time period corresponding to a duration of thebuffered data stream.
 3. The method of claim 2, further comprising:providing a signal at the interface in response to determining the voiceactivity is present; receiving an external clock signal at the interfaceof the microphone apparatus in response to providing the signal at theinterface, providing the multiplexed data stream at the interface afterreceiving the external clock signal.
 4. The method of claim 3, furthercomprising synchronizing the multiplexed data stream with the externalclock signal by synchronizing the buffered data stream with a firstportion of the external clock signal and synchronizing the real-timedata stream with a second portion of the external clock signal.
 5. Themethod of claim 1, wherein the generating the digital data streamincludes generating a first digital data stream from a first electricalsignal produced by a first acoustic sensor and generating a seconddigital data stream from a second electrical signal produced by a secondacoustic sensor, and wherein the providing the real-time data stream atthe interface includes providing a first real-time data stream based onthe first electrical signal and a second real-time data stream based onthe second electrical signal.
 6. The method of claim 5, furthercomprising synchronizing the first real-time data stream with a firstportion of a clock signal and synchronizing the second real-time datastream with a second portion of the clock signal.
 7. The method of claim1, further comprising providing the multiplexed data stream at a singleconnection of the interface of the microphone apparatus.
 8. The methodof claim 1, further comprising synchronizing the buffered data stream ofthe multiplexed data stream with a first portion of a clock signal andsynchronizing the real-time data stream of the multiplexed data streamwith a second portion of the clock signal.
 9. The method of claim 1,further comprising: operating the microphone apparatus in a first modebefore providing the multiplexed data stream at the interface of themicrophone apparatus; and operating the microphone apparatus in a secondmode while providing the multiplexed data stream at the interface of themicrophone apparatus, wherein a power consumption of the microphoneapparatus operating in the first mode is less than a power consumptionof the microphone apparatus operating in the second mode.
 10. Amicrophone apparatus comprising: a converter having an analog input anda digital output, wherein a digital data stream on the output of theconverter is produced from an electrical signal representative ofacoustic energy on the analog input of the converter; a buffer coupledto the digital output of the converter, wherein the buffer is configuredto buffer data representative of the electrical signal, the buffereddata having a delay time with respect to real-time data representativeof the electrical signal; a voice activity detector (VAD) coupled to thedigital output of the converter, wherein the VAD is configured to detectwhether voice activity is present in the electrical signal while data isbuffered in the buffer; and a multiplexer coupled to the buffer and tothe converter, the multiplexer coupled to an interface of the microphoneapparatus, wherein a multiplexed data stream is provided, by themicrophone apparatus, at the interface after voice activity is detected,the multiplexed data stream including a buffered data stream and areal-time data stream, the buffered data stream and the real-time datastream based on the electrical signal, the buffered data streamincluding non-real-time buffered data and real-time buffered data, aportion of the real-time buffered data of the buffered data streamtemporally overlapping a portion of the real-time data stream, andwherein the real-time data stream is provided, by the microphoneapparatus, at the interface after the buffered data stream is no longerprovided at the interface.
 11. The apparatus of claim 10, furthercomprising a local oscillator, wherein the microphone apparatus isclocked by the local oscillator at least before providing themultiplexed data stream at the interface.
 12. The apparatus of claim 11,wherein the apparatus is an integrated circuit.
 13. The apparatus ofclaim 11, further comprising an acoustic sensor packaged with theintegrated circuit, the acoustic sensor coupled to the analog input ofthe converter, the integrated circuit comprising at least the localoscillator, the converter, the buffer, the VAD, and the multiplexer. 14.The apparatus of claim 10, wherein the buffered data stream of themultiplexed data stream is synchronized with a first portion of a clocksignal and the real-time data stream of the multiplexed data stream issynchronized with a second portion of the clock signal.
 15. Theapparatus of claim 10, further comprising a controller coupled to theVAD and to the interface, the interface including a clock connection, adata connection, and a signal connection, wherein: a signal of thecontroller is provided at the signal connection of the interface inresponse to the VAD detecting voice activity, the microphone apparatusis clocked by an external clock signal provided at the clock connectionof the interface in response to the signal of the controller, themultiplexed data stream of the multiplexer is provided at the dataconnection of the interface in response to the external clock signal,and the real-time data stream is provided at the data connection of theinterface after the buffered data stream is no longer provided.
 16. Theapparatus of claim 15, wherein the buffered data stream is synchronizedwith a first edge of the external clock signal and the real-time datastream is synchronized with a second edge of the external clock signal.17. The apparatus of claim 10, wherein the digital data stream includesa first digital data stream generated from a first electrical signalproduced by a first acoustic sensor and a second digital data streamgenerated from a second electrical signal produced by a second acousticsensor, and wherein the real-time data stream includes a first real-timedata stream based on the first electrical signal and a second real-timedata stream based on the second electrical signal.
 18. The apparatus ofclaim 17, wherein the first real-time data stream is synchronized with afirst portion of a clock signal and the second real-time data stream issynchronized with a second portion of the clock signal.
 19. Theapparatus of claim 10, wherein the interface has a single data pin, andwherein the multiplexed data stream is provided at the single data pinof the interface after voice activity is detected.
 20. The apparatus ofclaim 10, wherein the microphone apparatus has a first mode of operationbefore the multiplexed data stream is provided at the interface, whereinthe microphone apparatus has a second mode of operation while themultiplexed data stream is provided at the interface, and wherein apower consumption of the first mode of operation is less than a powerconsumption of the second mode of operation.